In real-time electronic communication, an important measure of quality is the length of time between sending data and the beginning of the actual playback of the data. The delay between the transfer of data from a remote source and the rendering of data is often known as end-to-end communication latency. Communication latency may affect both one-way communication, such as server-to-terminal streaming, and multipoint communication, such as Internet telephony.
Latency may interfere with true real-time communication by requiring the receiver of a communication to delay playback of the communication. In addition, data buffers may be required to store an advance supply of data to compensate for momentary transmission or processing delays. Some non-zero latency, however, may be tolerable and even necessary in real-time communication systems. For example, a small latency may be useful to compensate for physical constraints of some communication systems. Optimizing latency generally provides a better communication experience for the user.
Latency may result from a variety of sources, including audio recording buffer delay, transmission delay, receiving data buffer delay, and playback delay. Various transient conditions experienced during the communication setup and start/stop operations in both half-duplex and full-duplex modes may cause the receiving buffer delay to differ from an optimal delay. Start/stop operations generally are necessary even for full-duplex operations in order to use available bandwidth efficiently. For example, some systems use a voice detection algorithm to determine when to start and stop sending voice data packets.
In real-time audio communications that use a packet-switched internet, network-introduced latency is generally variable and unpredictable. This results in difficulties in predetermining an optimal size for a receiving data buffer that accommodates all situations.
The determination of an optimal size for a receiving data buffer may be further complicated by a temporal drift that may exist between the asynchronous recording and playback devices operating at different recording/playback speeds. This temporal drift may cause either steady buildup or steady depletion of the receiving data buffer, which, in turn, may result in the actual latency in a communication deviating from an optimal latency.